asr-transcribe-to-text

name	asr-transcribe-to-text
description	Transcribes audio and video to speaker-labeled text: Qwen3-ASR full-audio transcription + whisper word timing + pyannote diarization, aligned into a who-said-what transcript BY DEFAULT (decoupled WhisperX-style — the audio is never cut before ASR). Handles local files, direct media URLs, and podcast/web pages; local MLX inference on macOS Apple Silicon or remote OpenAI-compatible ASR endpoints. Use when the user wants to transcribe recordings, podcasts, lectures, interviews, meetings, screen recordings, or any audio/video file; also use for ASR, Qwen ASR, speech-to-text, 转录, 语音转文字, 录音转文字, speaker diarization, who said what, 说话人分离, 说话人识别, 谁在说话 — speaker labels are the default, plain text is the opt-out. Also covers word-level timestamps via mlx-whisper for subtitles and audio-visual alignment (字幕, 时间戳, 音画对齐) and CAM++ voiceprint speaker identification. Also preprocesses: merging multi-segment recorder dumps (多段录音合并/拼接) and pitch-preserved speedup for metered-ASR quota uploads (飞书妙记/Feishu Minutes).
argument-hint	[audio-or-video-file-path-or-url ...]

ASR Transcribe to Text

Transcribe audio/video to speaker-labeled text. Default pipeline (decoupled, WhisperX-style): Qwen3-ASR transcribes the full audio with context intact, mlx-whisper supplies a word-level timing lattice, pyannote supplies speaker segments, and an aligner merges the three — the audio is never cut before ASR, so transcription quality stays at full-audio fidelity.

Mode	When	Speed	Cost
Local MLX	macOS Apple Silicon	15-27x realtime	Free
Remote API	Any platform, or when local unavailable	Depends on GPU	API/self-hosted

Configuration persists in ${CLAUDE_PLUGIN_DATA}/config.json.

Speaker labels are the default. Every run produces [start-end] SPEAKER_xx: text

• CSV. Plain-text-only output is the opt-out (--no-diarization) for monologues, podcasts, or when you just want a summary — see Step 3.

One-time setup for diarization: pyannote is a gated HuggingFace model — it needs a token once (## Speaker Diarization & Identification below). First run without it FAILS with setup steps; after setup, full capability is permanent and auto-detected.

Step 0: Detect Platform and Load Config

cat "${CLAUDE_PLUGIN_DATA}/config.json" 2>/dev/null

If config exists, read values and proceed to Step 1.

If config does not exist, auto-detect platform first:

python3 -c "
import sys, platform
is_mac_arm = sys.platform == 'darwin' and platform.machine() in ('arm64', 'aarch64')
print(f'Platform: {sys.platform} {platform.machine()}')
print(f'Apple Silicon: {is_mac_arm}')
if is_mac_arm:
    print('RECOMMEND: local-mlx')
else:
    print('RECOMMEND: remote-api')
"

Then use AskUserQuestion with platform-aware defaults:

For macOS Apple Silicon (recommended: local):

ASR setup — your Mac has Apple Silicon, so local transcription is recommended.

Q1: Transcription mode?
  A) Local MLX — runs on your Mac's GPU, no API key needed, 15-27x realtime (Recommended)
  B) Remote API — send audio to a server (vLLM, Tailscale workstation, etc.)

Q2: Does your network have an HTTP proxy that might intercept traffic?
  A) Yes — bypass proxy for ASR traffic (Recommended if using Shadowrocket/Clash)
  B) No — direct connection

For other platforms (recommended: remote):

ASR setup — local MLX requires macOS Apple Silicon. Using remote API mode.

Q1: ASR Endpoint URL?
  A) https://asr.example.com/v1/audio/transcriptions (Self-hosted remote ASR)
  B) http://localhost:8002/v1/audio/transcriptions (Local ASR server)
  C) Custom URL

Q2: Proxy bypass needed?
  A) Yes (Recommended for Shadowrocket/Clash/corporate proxy)
  B) No

Save config:

mkdir -p "${CLAUDE_PLUGIN_DATA}"
python3 -c "
import json
config = {
    'mode': 'MODE',           # 'local-mlx' or 'remote-api'
    'model': 'MODEL_ID',      # local: 'mlx-community/Qwen3-ASR-1.7B-8bit', remote: 'Qwen/Qwen3-ASR-1.7B'
    'max_tokens': 200000,     # local only, critical for long audio
    'endpoint': 'URL',        # remote only
    'noproxy': True,
    'max_timeout': 900        # remote only
    # 'diarization_declined': True  # set only after the user explicitly declines
    #   the pyannote setup in Step 3 — every run then warns + goes plain-text
    #   until an HF token appears (auto-detected)
}
with open('${CLAUDE_PLUGIN_DATA}/config.json', 'w') as f:
    json.dump(config, f, indent=2)
print('Config saved.')
"

Step 1: Resolve Input

Accept local files, direct media URLs, or web/podcast episode pages.

• Web or podcast page URL: inspect the page for an existing transcript first. Use an official/platform transcript only when it is directly accessible to the user's account. If the transcript endpoint requires a login token and none is available, say that clearly and fall back to ASR from the audio URL.
• Local file, direct media URL, or page URL fallback: run the bundled resolver. It extracts media from common page metadata (og:audio, media tags, JSON-LD, RSS-style enclosure links), downloads URLs with atomic temp-file replacement, verifies remote Content-Length when present, computes SHA-256, and validates the result with ffprobe.

uv run ${CLAUDE_PLUGIN_ROOT}/scripts/resolve_media_input.py \
  INPUT_FILE_OR_URL [INPUT_FILE_OR_URL2 ...] \
  --output-dir OUTPUT_DIR \
  --manifest OUTPUT_DIR/media_manifest.json

For suspicious or high-value downloads, add --decode-check to make ffmpeg decode the whole file before transcription:

uv run ${CLAUDE_PLUGIN_ROOT}/scripts/resolve_media_input.py \
  "https://www.xiaoyuzhoufm.com/episode/EPISODE_ID" \
  --output-dir OUTPUT_DIR \
  --manifest OUTPUT_DIR/media_manifest.json \
  --decode-check

Expected output:

Downloaded ... bytes in ...s -> OUTPUT_DIR/episode-title.m4a
OUTPUT_DIR/episode-title.m4a

Use the printed local path as INPUT_AUDIO in later steps. If ${CLAUDE_PLUGIN_ROOT} is empty, use the absolute path to this skill directory.

For third-party public podcasts or copyrighted media, save the transcript as a local file for the user's personal analysis. Do not paste a full long transcript into chat; provide a path, previews, summaries, or short excerpts instead.

Step 2: Extract Audio (if input is video)

For video files (mp4, mov, mkv, avi, webm), extract as 16kHz mono WAV:

ffmpeg -i INPUT_VIDEO -vn -acodec pcm_s16le -ar 16000 -ac 1 OUTPUT.wav -y

Audio files (wav, mp3, m4a, flac, ogg) can be used directly. Get duration:

ffprobe -v error -show_entries format=duration -of default=noprint_wrappers=1:nokey=1 INPUT_FILE

Cleanup: After transcription succeeds, delete extracted WAV files to save disk space.

Preprocess: Merge Segments & Shrink Metered Uploads (optional)

Run this BEFORE transcription when either applies:

• The recording is a multi-segment dump — body mics and field recorders split sessions into fixed-length files (e.g. TX02_MIC024_....wav, TX02_MIC025_....wav). Merge them and transcribe the merged file: full-audio context is the quality basis of the decoupled pipeline (Step 3), so transcribing segments separately throws away exactly what the architecture buys.
• The audio goes to a metered ASR (Feishu Minutes, any per-minute quota) — a pitch-PRESERVED speedup cuts billed duration directly, and modern ASR does not care: 1.3x was user-verified on Feishu Minutes (2026-07-16) with no perceptible recognition difference, and public Whisper benchmarks show no sharp WER drop until 2.0x (≤1.5x = safe zone, ~3% WER increase at 1.5x; >2x unusable).

Use the bundled script — it merges, normalizes to 16 kHz mono, optionally speeds up, and verifies its own output instead of trusting the ffmpeg exit code:

uv run ${CLAUDE_PLUGIN_ROOT}/scripts/prepare_asr_input.py SEG1.wav SEG2.wav -o merged.wav   # merge only
uv run ${CLAUDE_PLUGIN_ROOT}/scripts/prepare_asr_input.py SEG*.wav -o upload.m4a --speed 1.3  # merge + quota-saving speedup

Expected output:

Merge order:
  1. SEG1.wav  [pcm_s24le 48000Hz ch=1 1800.14s]
  2. SEG2.wav  [pcm_s24le 48000Hz ch=1 1800.15s]
[OK] duration: 4946.19s vs expected 4946.18s (delta +0.00s)
[OK] boundary 1 @ 1384.7s: max_volume -15.5 dB
[info] overall: mean_volume -38.3 dB, max_volume 0.0 dB
Wrote upload.m4a

• Segments sort by the YYYYMMDD_HHMMSS timestamp embedded in their filenames when every file has one (recorder dumps do); otherwise the given order is kept with a note — eyeball the printed merge order before transcribing.
• Self-verification: output duration must equal Σsegments ÷ speed (±1.5 s, hard FAIL otherwise); each splice gets a 10 s volume spot-check (dead air at a boundary = wrong order or a missing segment); overall loudness prints for comparison with the source.
• Speedup must be atempo-style pitch-preserved stretch — never sample-rate trickery, which shifts pitch and breaks both ASR accuracy and diarization voiceprints.
• Codec by extension: .m4a (AAC 48k — the metered-upload choice; ~30% smaller than mp3 at equal speech quality), .wav (pcm_s16le, local pipeline default), .flac (lossless archive master, ~50% of WAV), .mp3 (compatibility fallback only).
• Keep the originals until the transcript passes Step 4 verification.

Step 3: Transcribe (speaker labels by default)

Path A: Local MLX (macOS Apple Silicon) — default

Run the decoupled speaker pipeline — it handles dependency pins, model loading, and the critical max_tokens parameter internally. If ${CLAUDE_PLUGIN_ROOT} is empty, use the absolute path to this skill directory.

uv run ${CLAUDE_PLUGIN_ROOT}/scripts/speaker_transcribe.py \
  INPUT_AUDIO [INPUT_AUDIO2 ...] OUTPUT_DIR

Expected output (per file):

Device: mps
+ uv run .../transcribe_local_mlx.py ...        (leg 1: full-audio text)
+ uv run .../word_timestamps_whisper.py ...     (leg 2: timing lattice)
... diarization ...                             (leg 3: pyannote segments)
STEM: 42 turns, speakers=['SPEAKER_00', 'SPEAKER_01'], anchored_ratio=0.93
Wrote STEM.txt, STEM.csv, STEM.alignment.json

Outputs per input: <stem>.txt ([MM:SS - MM:SS] SPEAKER_xx + text), <stem>.csv (file,start,end,duration,speaker,text — feeds review UIs and voiceprint ID), <stem>.diarization.json, <stem>.alignment.json (provenance

• anchored_ratio trust signal; < 0.5 prints a loud warning — verify labels against the audio before trusting them). Intermediate legs are cached in OUTPUT_DIR/_align/ so re-runs are cheap (--force redoes them).

Before a long first run, smoke-test the Qwen3 leg once:

uv run ${CLAUDE_PLUGIN_ROOT}/scripts/transcribe_local_mlx.py --smoke-test

Expected output includes Dependency stack: mlx-audio 0.3.1, mlx-lm 0.30.5, transformers 5.0.0rc3 and Smoke test OK. For performance details and the max_tokens truncation issue, see references/local_mlx_guide.md.

How it works (and why): full-audio Qwen3-ASR text + mlx-whisper word timestamps + pyannote speaker segments, aligned after the fact — the audio is never cut before transcription, so ASR keeps full context. Architecture, alignment algorithm, and failure modes: references/decoupled_speaker_alignment.md.

First run: pyannote needs a one-time HuggingFace token. If the script exits with the setup hint (exit code 3), STOP and use AskUserQuestion:

Speaker diarization needs a one-time setup (gated model, free):
  1. Accept terms at https://hf.co/pyannote/speaker-diarization-3.1
  2. Run `huggingface-cli login` (or set HF_TOKEN)

Options:
A) Set it up now — I'll wait, then rerun with full speaker labels (Recommended)
B) Continue without speakers this time — plain text only

• A → after the user confirms login, rerun the same command. The token is auto-detected every run; full capability is permanent from then on.
• B → persist the choice (diarization_declined: true in config.json) and rerun the SAME command. The script detects the flag, prints a one-line warning with the two setup steps, and auto-falls back to plain text for that run — no need to pass --no-diarization (the fallback is automatic now, enforced in the script not just the doc). The same warn-and-continue happens on every later run while the token is still missing. When a token later appears, diarization resumes automatically (the flag is ignored once a token is present) — mention this so the user knows setup is all that's needed.

Plain-text fast path (monologue, podcast, "just summarize it"):

uv run ${CLAUDE_PLUGIN_ROOT}/scripts/speaker_transcribe.py \
  INPUT_AUDIO OUTPUT_DIR --no-diarization

Remote/pre-made ASR text (e.g. from Path B, or another ASR service): skip the Qwen3 leg and align that text instead. --text-file pairs ONE transcript with ONE input wav — passing multiple inputs is rejected (one transcript can't be aligned to several files):

uv run ${CLAUDE_PLUGIN_ROOT}/scripts/speaker_transcribe.py \
  INPUT_AUDIO OUTPUT_DIR --text-file TRANSCRIPT.txt

Non-Apple-Silicon machines: the whisper timing leg is MLX-only. Without it there is no timing lattice to align speakers onto — run with --no-diarization and tell the user speaker mode currently requires Apple Silicon (cloud ASR with built-in diarization, e.g. Feishu Minutes, is the no-local-GPU alternative).

Before batching many short files (promo clips, montage cuts — anything that may contain music-only audio), read ## Batch Transcription (many short files) below: one music-only clip can stall the whole batch for 10+ minutes.

Path B: Remote API

The remote endpoint returns plain text only — speakers are added locally by aligning that text (leg 1) with the local timing + diarization legs. So Path B = fetch text remotely, then run Path A's pipeline with --text-file.

Health check first (skip if already verified this session):

python3 -c "
import json, subprocess, sys
with open('${CLAUDE_PLUGIN_DATA}/config.json') as f:
    cfg = json.load(f)
base = cfg['endpoint'].rsplit('/audio/', 1)[0]
noproxy = ['--noproxy', '*'] if cfg.get('noproxy', True) else []
result = subprocess.run(
    ['curl', '-s', '--max-time', '10'] + noproxy + [f'{base}/models'],
    capture_output=True, text=True
)
if result.returncode != 0 or not result.stdout.strip():
    print(f'HEALTH CHECK FAILED: {base}/models', file=sys.stderr)
    sys.exit(1)
print(f'Service healthy: {base}')
"

Read config and send via curl:

python3 -c "
import json, subprocess, sys, os, tempfile
with open('${CLAUDE_PLUGIN_DATA}/config.json') as f:
    cfg = json.load(f)
noproxy = ['--noproxy', '*'] if cfg.get('noproxy', True) else []
timeout = str(cfg.get('max_timeout', 900))
audio_file = 'AUDIO_FILE_PATH'
output_json = tempfile.mktemp(suffix='.json', prefix='asr_')

result = subprocess.run(
    ['curl', '-s', '--max-time', timeout] + noproxy + [
        cfg['endpoint'],
        '-F', f'file=@{audio_file}',
        '-F', f'model={cfg[\"model\"]}',
        '-o', output_json
    ], capture_output=True, text=True
)

with open(output_json) as f:
    data = json.load(f)
if 'text' not in data:
    print(f'ERROR: {json.dumps(data)[:300]}', file=sys.stderr)
    sys.exit(1)
text = data['text']
print(f'Transcribed: {len(text)} chars', file=sys.stderr)
print(text)
os.unlink(output_json)
" > OUTPUT.txt

Then attach speakers locally (Apple Silicon + pyannote token required):

uv run ${CLAUDE_PLUGIN_ROOT}/scripts/speaker_transcribe.py \
  INPUT_AUDIO OUTPUT_DIR --text-file OUTPUT.txt

If remote health check fails, diagnose in order:

1. Network: ping -c 1 HOST or tailscale status | grep HOST
2. Service: tailscale ssh USER@HOST "curl -s localhost:PORT/v1/models"
3. Proxy: retry with --noproxy '*' toggled

Step 4: Verify Output

After transcription, check for truncation — the most common failure mode:

1. Confirm output is not empty
2. Check character count is plausible (~400 chars/min for Chinese, ~200 words/min for English)
3. Check the ending — does it trail off mid-sentence? If so, max_tokens was exhausted
4. Show user the first and last ~200 characters as preview
5. Speaker path: check the alignment report — anchored_ratio should be ≥ 0.5 (the script warns when lower), the speaker count should be plausible for the recording (a two-person interview showing 5 speakers, or a monologue split into 2+, means diarization over-segmented — see references/speaker_diarization.md for when to distrust labels)

If truncated or wrong, use AskUserQuestion:

Transcription may be truncated:
- Expected: ~[N] chars for [M] minutes of audio
- Got: [actual] chars ([pct]% of expected)
- Last line: "[last 100 chars...]"

Options:
A) Retry with higher max_tokens (current: [N], try: [N*2])
B) Switch mode — try [local/remote] instead
C) Save as-is — the output looks complete to me
D) Abort

Step 5: Fallback — Overlap-Merge (Remote API Only)

If single remote request fails (timeout, OOM), fall back to chunked transcription:

python3 ${CLAUDE_PLUGIN_ROOT}/scripts/overlap_merge_transcribe.py \
  --config "${CLAUDE_PLUGIN_DATA}/config.json" \
  INPUT_AUDIO OUTPUT.txt

Splits into 18-minute chunks with 2-minute overlap, merges using punctuation-stripped fuzzy matching. See references/overlap_merge_strategy.md for algorithm details.

For local MLX mode, overlap-merge is unnecessary — the bundled script handles chunking internally with max_tokens=200000.

Step 6: Recommend Transcript Correction

ASR output always contains recognition errors — homophones, garbled technical terms, broken sentences. After successful transcription, proactively suggest running the transcript-fixer skill on the output:

Transcription complete: [N] chars saved to [output_path].

ASR output typically contains recognition errors (homophones, garbled terms, broken sentences).
Would you like me to run /daymade-audio:transcript-fixer to clean up the text?

Options:
A) Yes — run daymade-audio:transcript-fixer on the output now (Recommended)
B) No — the raw transcription is good enough for my needs
C) Later — I'll run it myself when ready

If the user chooses A, invoke the transcript-fixer skill with the output file path. The two skills form a natural pipeline: transcribe → correct → review.

Reconfigure

rm "${CLAUDE_PLUGIN_DATA}/config.json"

Then re-run Step 0.

Batch Transcription (many short files)

Passing many files to one transcribe_local_mlx.py invocation is efficient (model loads once) — but only when every file contains actual speech. If the batch may include music-only / BGM-only clips (short promo videos, montage clips with subtitles instead of voiceover), do NOT batch them in one process:

• On music/rhythm-only audio the model can fall into a repetition loop hallucination (e.g. endless "One, two, three, one, two, three...") that burns toward max_tokens=200000 — one such file can stall for 10+ minutes and starve the whole batch.
• Drive batch jobs one-file-per-process with a per-file timeout (e.g. timeout 240 / perl -e 'alarm 240; exec @ARGV' around each invocation, skip on timeout, second pass for failures). A stuck file then costs 4 minutes, not the batch.
• For a stuck file, retry with --max-tokens 3000: real speech in a short clip fits comfortably; a looping file gets truncated output you can classify.
• Detect "no speech" instead of shipping garbage: if the transcript's unique-word ratio is extremely low (e.g. len(set(words))/len(words) < 0.06 on a 40+ char output), the clip almost certainly has no voiceover — label it as such rather than delivering the loop text. (Downstream OCR of on-screen captions is the actual fix for subtitle-only videos.)

Word-Level Timestamps (subtitles, audio-visual alignment)

mlx-whisper's word timing is now the timing leg of the default speaker pipeline (leg 2 — scripts/word_timestamps_whisper.py runs it automatically). This section is for using word timestamps STANDALONE: subtitle generation, aligning narration to shot boundaries, per-clip captioning.

Qwen3-ASR is an LLM-decoder ASR: it emits plain text with no alignment information, on both local and remote paths. When the task needs to know when each word is spoken, use mlx-whisper with word_timestamps=True. Whisper's cross-attention word alignment is the de-facto local solution for this class of task.

Key facts (full recipe in references/whisper_word_timestamps.md):

• Model: mlx-community/whisper-large-v3-turbo (~1.6GB). Its Chinese WER is higher than Qwen3-ASR for pure transcription, but for alignment tasks Qwen3-ASR is not an option at all; prime domain terms via initial_prompt.
• Segment granularity trap: on short videos (15–40s) whisper often returns the whole clip as one segment — always work from the word list and assign words to time windows by midpoint.
• Pairs with ffmpeg scene detection (select='gt(scene,0.3)') for the visual side; avoid PySceneDetect on non-ASCII paths.

Speaker Diarization & Identification (who said what)

Speaker labels are the DEFAULT output of Step 3 (decoupled architecture: full-audio Qwen3-ASR text + whisper timing lattice + pyannote segments, aligned — never cut-then-transcribe). This section covers the pieces.

• The pipeline — scripts/speaker_transcribe.py runs all three legs + alignment in one command and writes the speaker-labeled transcript + CSV. Architecture, alignment algorithm, trust signals (anchored_ratio), and failure modes: references/decoupled_speaker_alignment.md. Production pitfalls (over-segmentation, mic-domain effects, when to distrust labels): references/speaker_diarization.md.
• Diarization alone — scripts/diarize_speakers.py emits just the speaker × time segments (no transcription).
• Legacy cascade — scripts/speaker_transcribe_cascade.py is the old cut-then-transcribe variant (diarize → slice audio per turn → ASR each slice). It breaks ASR context at every cut and lowers text quality; kept only for extremely noisy / heavy-overlap audio where per-slice isolation of a dominant near-field speaker beats full-audio ASR. Everything else uses the decoupled default.
• Voiceprint identification — diarization labels are anonymous (SPEAKER_00…) and per-file. To map them to real names, unify a speaker across files, or collapse diarization's over-segmentation, use CAM++ voiceprints via scripts/voiceprint_id.py. Recipe and the critical acoustic-domain caveat — a voiceprint built from one mic type matches the same person on a different mic far less well: references/voiceprint_speaker_id.md.

One-time pyannote setup (gated model): accept terms at hf.co/pyannote/speaker-diarization-3.1, then huggingface-cli login once (or set HF_TOKEN). Auto-detected on every run afterward.

Transcript Audit & Review (HTML)

After diarization you get a CSV per file (file,start,end,duration,speaker,text). The bundled audit HTML generator turns those CSVs into a single, reader-first review page with audio playback, per-turn flags/notes, speaker aliasing, and export.

Generate it from a speaker-transcribe output directory (run from the skill directory, or replace <path-to-asr-transcribe-to-text> with the actual install path):

uv run scripts/generate_audit_html.py \
  OUTPUT_DIR \
  --output OUTPUT_DIR/audit/index.html \
  --audio-dir /path/to/original/audio

Defaults assume a flat layout under PROJECT_DIR: PROJECT_DIR/*.csv transcripts, PROJECT_DIR/*.diarization.json, and the original audio files placed next to the outputs. speaker_transcribe.py itself writes the CSV, TXT, and diarization files flat under its OUTPUT_DIR. If your project uses a different structure, override any of those paths:

uv run <path-to-asr-transcribe-to-text>/scripts/generate_audit_html.py \
  /path/to/project \
  --output /path/to/project/audit/index.html \
  --csv-dir /path/to/project/csv \
  --txt-dir /path/to/project/txt \
  --diarization-dir /path/to/project/diarization \
  --audio-dir /path/to/project/audio \
  --original-dir /path/to/project/original \
  --manifest /path/to/project/manifest.json \
  --title "Project Audit" \
  --subtitle "Speaker-labeled transcript review" \
  --storage-key "project-audit" \
  --known-speaker "Speaker A" \
  --known-speaker "Speaker B"

Key CLI options:

Option	Meaning
project_dir	Base project directory (required)
--output	Where to write index.html
--csv-dir	Directory containing *.csv transcript files
--txt-dir	Directory containing *.txt plain-text transcripts (optional)
--diarization-dir	Directory containing *.diarization.json files
--audio-dir	Directory containing playback audio files
--original-dir	Directory containing original source media (optional)
--manifest	JSON manifest mapping file IDs to metadata (optional)
--title / --subtitle	Page title and subtitle
--storage-key	localStorage namespace for state persistence
--known-speaker	Repeatable; "Name" auto-assigns a color, "Name=#hex" sets one explicitly
--material-final / --material-rough	Repeatable material classification labels used for filtering

The output is a single self-contained HTML file with no external dependencies. Open it in a browser to review, flag, and annotate turns; the export button produces a report of all flagged rows with reasons and notes.

Troubleshooting

Local MLX fails while loading the model

If model loading fails with an error like:

AttributeError: 'str' object has no attribute '__module__'

the agent is probably using an unpinned or stale copy of the local MLX script. The known-good stack is:

mlx-audio 0.3.1
mlx-lm 0.30.5
transformers 5.0.0rc3

Run the bundled --smoke-test command and confirm the dependency stack line matches. Do not start a long transcription until the smoke test succeeds.

${CLAUDE_PLUGIN_ROOT} is empty

Some runtimes do not set skill environment variables. Use the absolute path to the skill directory that contains this SKILL.md, then run scripts/transcribe_local_mlx.py from there.

Bundled Resources

Scripts:

• resolve_media_input.py — Resolve local paths, direct media URLs, and podcast/web pages into validated local media files
• prepare_asr_input.py — Merge multi-segment recordings + normalize for ASR (16 kHz mono), optional pitch-preserved speedup for metered uploads; self-verifies duration math and splice boundaries
• transcribe_local_mlx.py — Local MLX transcription (macOS ARM64, PEP 723 deps)
• speaker_transcribe.py — DEFAULT pipeline: decoupled multi-speaker transcription (full-audio Qwen3-ASR + whisper word timing + pyannote diarization, aligned) → speaker-labeled transcript + CSV; --no-diarization plain-text fast path; --text-file for remote/pre-made ASR text
• align_speakers.py — Decoupled alignment core (stdlib): maps full transcript onto whisper word lattice + pyannote segments; usable standalone for debugging
• word_timestamps_whisper.py — mlx-whisper word-level timestamps → JSON timing lattice (Apple Silicon)
• speaker_transcribe_cascade.py — LEGACY cut-then-transcribe variant (extremely noisy / heavy-overlap audio only)
• diarize_speakers.py — Speaker diarization alone (pyannote 3.1 @ MPS) → per-segment JSON
• voiceprint_id.py — CAM++ voiceprint enroll/match: map anonymous SPEAKER_xx to real names
• overlap_merge_transcribe.py — Chunked transcription with overlap merge (remote API fallback)
• generate_audit_html.py — Build a self-contained HTML audit/review page from speaker-transcribe CSV outputs

References:

• decoupled_speaker_alignment.md — The default architecture: why decouple, alignment algorithm, trust signals, failure modes
• speaker_diarization.md — Production pitfalls: over-segmentation, mic-domain effects, when to distrust labels; legacy cascade notes
• voiceprint_speaker_id.md — CAM++ speaker ID: enroll/match, threshold+margin gates, the acoustic-domain caveat, bootstrap
• local_mlx_guide.md — Performance benchmarks, max_tokens truncation, model compatibility
• whisper_word_timestamps.md — mlx-whisper word timing: the timing leg of the default pipeline; standalone subtitle/AV-alignment recipe
• overlap_merge_strategy.md — Why naive chunking fails, fuzzy merge algorithm